I have a Polycom IP 501 (a few actually). When I call an IP501 to IP501 within the office, using Asterisk 1.0.9 and SIP, my voice level is bearable - like normal PSTN, when listening to myself and my calling party.
But,when I use asterisk to use the TDM400P card and its FXO port to call to outside world, my voice is loader than the voice of whom I am calling and slightly delayed. I start to slur my speech as I can only hear myself 
Is there a setting in zapata.conf for this, or is it my phone (although SIP to SIP is OK); Is there an adjestment in extensions.conf.
In other words, how can I turn down my own voice monitoring when using a Sip to PSTN connection using the TDM400P FXO.
or should i say
IN OTHER WORDS HOW CAN I TURN …
Thanks,